Commit graph

508 commits

Author SHA1 Message Date
Sebastian Dröge
d24e68719b rtspsrc: Don't propagate spurious state change returns from internal elements further
We handle them inside rtspsrc and override them in all other cases anyway, so
do the same for "internal" state changes like PAUSED->PAUSED and
PLAYING->PLAYING.

This keeps unexpected NO_PREROLL to confuse state changes in GstBin.

See also https://bugzilla.gnome.org/show_bug.cgi?id=760532

https://bugzilla.gnome.org/show_bug.cgi?id=765689
2016-04-27 20:52:15 +03:00
Aleix Conchillo Flaqué
c36930535d rtspsrc: add srtp rollover counters from mikey crypto sessions
The server can send multiple crypto sessions, one for each SSRC with its
own rollover counter. We parse this information and pass it to the SRTP
decoder via the "request-key" signal.

https://bugzilla.gnome.org/show_bug.cgi?id=730540
2016-04-15 18:12:06 +02:00
Sebastian Dröge
df247f091c rtpjitterbuffer: Add RFC7273 media clock handling
https://bugzilla.gnome.org/show_bug.cgi?id=762259
2016-04-03 11:24:34 +03:00
Vineeth TM
1071309870 good: use new gst_element_class_add_static_pad_template()
https://bugzilla.gnome.org/show_bug.cgi?id=763076
2016-03-24 14:32:20 +02:00
Jihae Yi
da5c8a954c rtspsrc: avoid potentially overflowing expression
https://bugzilla.gnome.org/show_bug.cgi?id=757569
2016-03-24 14:28:50 +02:00
Sebastian Dröge
bf5a72a6dd rtspsrc: Properly error out if binding the UDP sockets fails
udpsrc is not returning us a socket in that case.
2016-02-28 13:01:34 +02:00
Aleix Conchillo Flaqué
665d14a2a0 rtspsrc: handle rtcp/srtcp caps properly when using interleaved data
We check the stream profile and use the proper RTCP caps:
application/x-srtcp if we are using a secure profile and
application/x-rtcp otherwise.

https://bugzilla.gnome.org/show_bug.cgi?id=760556
2016-01-18 11:29:25 +02:00
Hyunjun Ko
3300039513 rtspsrc: replace duplicated codes to call new base sdp apis
https://bugzilla.gnome.org/show_bug.cgi?id=745880
2015-12-31 17:12:09 +02:00
Evan Callaway
4718870959 rtspsrc: Retry connection if tunneling needs authentication
Leverage response from gst_rtsp_connection_connect_with_response to
determine if the connection should be retried using authentication.  If
so, add the appropriate authentication headers based upon the response
and retry the connection.

https://bugzilla.gnome.org/show_bug.cgi?id=749596
2015-12-14 16:42:27 +01:00
Luis de Bethencourt
4735d2a9a5 rtspsrc: check port-range format
The string could exist but with a wrong format, in that case we still want
to reset the values of client_port_range.min and max like we do if there is
no string.

CID 1139593
2015-12-14 14:53:57 +00:00
Reynaldo H. Verdejo Pinochet
a4c8ec8bd7 rtspsrc: remove unnecessary NULL checks before g_free() 2015-11-15 01:43:08 -08:00
Sebastian Dröge
01c0f8723f rtpbin/rtpjitterbuffer/rtspsrc: Add property to set maximum ms between RTCP SR RTP time and last observed RTP time
https://bugzilla.gnome.org/show_bug.cgi?id=755125
2015-09-25 23:55:05 +02:00
Jan Schmidt
b38e24995b rtspsrc: Don't parse key data more than needed.
When an auxilliary streams are present in the SDP media,
there's no need to re-parse the SDP attributes multiple
times.
2015-09-05 23:44:51 +10:00
Jan Schmidt
fe4ed1d1df rtspsrc: Fix SRTP + RTX, auth access, a leak, and an invalid memory access.
In parse_keymgmt(), don't mutate the input string that's been passed
as const, especially since we might need the original value again if
the same key info applies to multiple streams (RTX, for example).

When a resource is 404, and we have auth info - retry with the auth
info the same as if we had receive unauthorised, in case the resource
isn't even visible until credentials are supplied.

Fix a memory leak handling Mikey data.

When generating a random keystring, don't overrun the 30 byte
buffer by generating 32 bytes into it.
2015-09-05 23:44:51 +10:00
Vineeth TM
ba8cda54f4 rtspsrc: Trivial fix to check correct condition
When checking for describe method, because of missing parentheses, wrong
condition is being checked, which will result in wrong behavior.

https://bugzilla.gnome.org/show_bug.cgi?id=753912
2015-08-21 11:06:57 +03:00
George Kiagiadakis
e2f2f087ec rtspsrc: send the User-Agent header
Sometimes it is useful to know this information on the
server side. Other popular implementations (vlc, ffmpeg, ...)
also send this header on every message.

This includes a new "user-agent" property that the user
can set to use a custom User-Agent string. The default
is "GStreamer/<version>"

https://bugzilla.gnome.org/show_bug.cgi?id=750101
2015-08-14 15:59:06 +02:00
George Kiagiadakis
af03341e26 rtspsrc: wrap gst_rtsp_message_init_request in a local function
This will allow adding common request initialization, like the
user agent string, in just one place.
2015-08-14 15:59:06 +02:00
Vineeth TM
cf19525d5c rtspsrc: assertion error due to wrong condition check
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j

https://bugzilla.gnome.org/show_bug.cgi?id=753009
2015-07-30 15:51:25 +03:00
Sebastian Dröge
39a90710b7 rtspsrc: Strip keys from the fmtp that we use internally in our caps
Skip keys from the fmtp, which we already use ourselves for the
caps. Some software is adding random things like clock-rate into
the fmtp, and we would otherwise here set a string-typed clock-rate
in the caps... and thus fail to create valid RTP caps

https://bugzilla.gnome.org/show_bug.cgi?id=753009
2015-07-29 14:31:49 +01:00
Hyunjun Ko
f560a3d223 rtspsrc: Add new ntp-time-source property and deprecate use-pipeline-clock property
Enable to use new ntp-time-source property of rtpbin

https://bugzilla.gnome.org/show_bug.cgi?id=751496
2015-06-25 17:16:49 +02:00
Xavier Claessens
b0b3e8e2cc rtspsrc: Add a GTlsInteraction property
It can be used for TLS client authentication.

https://bugzilla.gnome.org/show_bug.cgi?id=750471
2015-06-09 20:03:18 -04:00
Sebastian Dröge
0a7823b30f rtspsrc: Set RTP profile on the rtpsession objects
https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
9865730cfa rtspsrc: Fix up last commit 2015-05-04 16:50:38 +02:00
Sebastian Dröge
d08f488598 rtspsrc: Only do RTX when using a feedback profile 2015-05-04 16:47:30 +02:00
Sebastian Dröge
33693525b9 rtspsrc: Only enable retransmissions if there is retransmission info in the SDP
Otherwise we're going to send early RTCP and NACKs in non-feedback sessions
too, which will confuse servers.

https://bugzilla.gnome.org/show_bug.cgi?id=748627
2015-04-29 15:53:09 +02:00
Luis de Bethencourt
671b4d25cd remove unused enum items PROP_LAST
This were probably added to the enums due to cargo cult programming and are
unused. Removing them.
2015-04-24 17:01:12 +01:00
Jesper Larsen
3528046773 rtspsrc: Fix RTCP caps leak
https://bugzilla.gnome.org//show_bug.cgi?id=748353
2015-04-23 14:56:27 +01:00
Nicolas Dufresne
84725d62b5 rtspsrc: Fix segment in TCP mode
It is expected that buffers are time-stamped with running time. Set
a segment accordingly. In this case we pick 0,-1 as this is what udpsrc
would do. Depayloaders will update the segment to reflect the playback
position.

https://bugzilla.gnome.org/show_bug.cgi?id=635701
2015-03-26 17:54:08 -04:00
Nicolas Dufresne
32aed67144 rtspsrc: Remove useless function
This function didn't do anything special, let's not use a function for
that.
2015-03-25 15:28:24 -04:00
Nicolas Dufresne
8afc8c8f3b rtspsrc: Fix seeking query
The segment start/stop in the query is meant to represent the seekable
portion of the stream. It does not match the segment start/stop. Instead
export 0 to duration.
2015-03-24 16:51:12 -04:00
Nicolas Dufresne
1c27002ebd rtspsrc: Improve trace readability
Change the command number into strings.
2015-03-18 17:32:36 -04:00
Ramiro Polla
90be7b4e1e rtspsrc: properly escape percent sign in documentation 2015-03-14 14:22:39 +00:00
Linus Svensson
398296d978 rtspsrc: Don't include payload type in the caps for framesize
When the sdp media attribute framesize are converted to caps
the <payload> should not be included.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
2015-03-09 10:18:35 +01:00
Sebastian Dröge
c34a7cb90d rtspsrc: Fix handling of interleaved (TCP) streams
We need to set up the transport in any case, not just if we have a container
stream or a non-interleaved stream. Only if we have an interleaved stream and
are retrying, we should not set up the stream again.

https://bugzilla.gnome.org/show_bug.cgi?id=745599
2015-03-05 12:15:04 +01:00
Sebastian Dröge
b4aaa11f97 rtspsrc: Don't unref caps we don't own 2015-03-05 09:56:37 +01:00
Sebastian Dröge
297d808acc rtspsrc: Push RTCP caps on the RTCP pads
Otherwise we will get not-negotiated later from rtpbin, and will never be able
to send RTCP packets back to the server. Note that error flow returns from the
RTCP pads are ignored, that's why it didn't fail more visible before.
2015-03-05 09:47:29 +01:00
Sebastian Dröge
788074733c rtspsrc: Make sure to send SEGMENT events on all pads 2015-03-05 09:47:29 +01:00
Krzysztof Kotlenga
e3ca4d1c86 rtspsrc: improve error message when unauthorized
Make use of NOT_AUTHORIZED error code instead of falling back to generic
READ error.

https://bugzilla.gnome.org/show_bug.cgi?id=601733
2015-02-24 11:08:27 +02:00
Tim-Philipp Müller
603c1d71a1 rtspsrc: fix awkward if clause 2015-02-08 12:03:10 +00:00
Wim Taymans
852c040c89 rtspsrc: fix container handling
We detect a container correctly now so we need to revert the weird
check there was before.
Use gst_rtspsrc_stream_push_event() to push the caps event on the
right pad.

See https://bugzilla.gnome.org/show_bug.cgi?id=739391
2015-02-03 17:39:10 +01:00
Sebastian Dröge
d5aab81a77 Constify some static arrays everywhere 2015-01-21 09:55:53 +01:00
Aleix Conchillo Flaqué
07c5d1820a rtspsrc: set PLAYING state after configuring caps
We set to PLAYING after we have configured the caps, otherwise we
might end up calling request_key (with SRTP) while caps are still
being configured, ending in a crash.

https://bugzilla.gnome.org/show_bug.cgi?id=740505
2014-12-31 12:49:11 +00:00
Sebastian Dröge
d18b893d28 rtspsrc: Don't dereference NULL if a suitable stream for the AUX element can't be found
CID 1258717
2014-12-18 11:51:12 +01:00
Sebastian Dröge
6b2fc2de8d rtspsrc: Add something to the debug logs if an RTX AUX element can't be added
... because the application already has a signal handler set up here.
2014-12-16 16:40:08 +01:00
Matthew Waters
bf0a19bf02 rtspsrc: add retransmission support according to RFC4588
Based on the client-rtpaux example
2014-12-16 16:40:08 +01:00
Aleix Conchillo Flaqué
00ca83629b rtspsrc: fix leak for mikey base64 decoded key-mgmt
https://bugzilla.gnome.org/show_bug.cgi?id=740392
2014-11-20 09:15:56 +01:00
Aleix Conchillo Flaqué
d15ebcbf62 rtspsrc: mikey related memory leaks
https://bugzilla.gnome.org/show_bug.cgi?id=739430
2014-10-31 10:03:47 +00:00
Aleix Conchillo Flaqué
bd392d72ee rtspsrc: set full stream caps on internal src TCP pads
Set the complete stream caps on the TCP internal src pads. Otherwise,
ptdemux will not properly detect the caps change.

https://bugzilla.gnome.org/show_bug.cgi?id=737868
2014-10-21 11:33:01 +02:00
Sebastian Dröge
d1c7f2e4d1 rtspsrc: Fix compiler warnings
gstrtspsrc.c:7939:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
      'GstRTSPResult' [-Werror,-Wenum-conversion]
    res = gst_sdp_message_new (&sdp);
        ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~
gstrtspsrc.c:7944:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
      'GstRTSPResult' [-Werror,-Wenum-conversion]
    res = gst_sdp_message_parse_uri (uri, sdp);
        ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
2014-09-26 13:46:16 +03:00
Sanjay NM
f62076e49c rtsp: clarify expression so operator precedence is clear
https://bugzilla.gnome.org/show_bug.cgi?id=736903
2014-09-24 00:48:09 +01:00